Communication of multimedia data requires optimum performance in several parameters in order to provide the data fast enough to preserve the quality of the multimedia services (quality of service—QoS) being offered, such as VoIP, audio distribution or video distribution. The data rate, i.e., the rate of signaling each data bit or symbol, has to be fast enough to provide data at the rate or faster than the rate of consumption by the receiving device's application. In packet data communications protocols, the data rate is further complicated by the delay between sending the packets of data. The delay in delivering an individual packet of data is the packet latency. The variation of that delay across multiple receive packets is called jitter. In highly congested shared networks, devices use various methods to contend for access to the network so their packets can be sent with low enough latency and jitter in order to ensure the necessary QoS requirements of the data-consuming application.
In some communications protocols, for example HomePlug® AV and AV2, which are hereby incorporated by reference, a “regular” MAC protocol data unit (MPDU) can be transmitted to a receiving node's physical layer (PHY) of the OSI model and receive an acknowledgement (ACK) back for each successfully transmitted MPDU. These protocols also support a burst mode which allows the transmitter to transmit multiple long MPDUs without relinquishing the medium, and before soliciting a response. The response, a selective acknowledgement (SACK) from the receiver back to the transmitter, provides the reception status for all of the MPDU's sent by the transmitting PHY to the receiver's PHY. Long MPDUs in burst mode are separated by burst interfame spacing (BIFS). Because MPDU bursts only require a single SACK response, the time to send packets and get ACK responses is reduced and the protocol efficiency increases for that communications exchange. In the burst mode, the start of frame (SOF) delimiter contains a counter field (MPDUCnt) that indicates how many MPDUs follow the current MPDU (with the value “0” indicating the last MPDU in the sequence. FIG. 1 shows an example of MPDU bursting as known to those skilled in the art.
The protocols cited above also allow for bidirectional bursting. In this mode, the transmitter allows part of the time it reserved to burst data to the receiver for the receiver to send data back to the original transmitter. It serves as an effective back channel that does not need to be negotiated with the network. The receiving station initiates bi-directional bursting by sending a “request reverse transmission flag” (RRTF) and “request reverse transmission length” (RRTL) fields in the frame control section of the SACK. The RRTL field specifies the minimum required frame length for the Reverse SOF (RSOF) MPDU. Upon receiving the request, the original transmitter decides whether to honor the request and the duration. Obviously, if the request of for more time than the original transmitter has reserved, it will be denied. FIG. 2 illustrates an example of the bidirectional burst mechanism as known to those skilled in the art. When the receiver (Dev B) determines that it wants to transmit in the reverse direction, it sets the RRTF and RRTL fields in the SACK or RSOF. This is set until the original transmitter (Dev A) responds, granting the request for the maximum duration, or until there is no longer a need to request a transmission in the reverse direction. FIG. 3 shows the various interframe spaces during a bidirectional burst. These spaces result in increased latency (reduced efficiency).
Multicast transmissions do not allow individual receivers to acknowledge that data was received by each receiver. This is not acceptable for isochronous systems that require specific levels of QoS for each device. Some protocols try to get around this limitation by allowing one station in the group to act as the proxy for the others but it does not provide information about all of the devices in the multicast. In a busy network there may not be enough time or bandwidth to accommodate the QoS requirements for isochronous traffic. A better method is needed to improve packet latency and hence communications efficiency in congested multimedia networks.